Arnd Vehling wrote:
Hi,
i am having severe problems with asterisk svn trunk. SIP/RTP is pretty
unreliable. Calls between 2 phones connected directly (sip) to the box
always fail to establish a correct rtp stream. Looks like an NAT issue
because the rtp stream failing/not getting setup is the one to the phone
behind a NAT box. NAT is setup correct though. Works with older asterisk
version.
better yet, I get a nice crasher as soon as I try to call out via SIP
(see bug http://bugs.digium.com/view.php?id=7854 )
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